U.S. patent number 7,460,495 [Application Number 11/066,137] was granted by the patent office on 2008-12-02 for serverless peer-to-peer multi-party real-time audio communication system and method.
This patent grant is currently assigned to Microsoft Corporation. Invention is credited to Jin Li.
United States Patent |
7,460,495 |
Li |
December 2, 2008 |
Serverless peer-to-peer multi-party real-time audio communication
system and method
Abstract
A serverless peer-to-peer (P2P) multi-party real-time audio
communication system and method in which each of the peers takes a
turn mixing and compressing the audio and redelivering the
compressed audio. An input audio stream is divided or split into
frames. At each frame, one peer node is selected to mix and
redeliver the audio to the remainder of the peers in the network.
The number of frames mixed and redelivered by a certain peer is
proportional to its available resources (such as the upload
bandwidth or computational power). The P2P audio communication
system and method flexibly balances the load of the peers, such
that peers having more resources are able to assist peers having
fewer resources. This enables the P2P audio communication system
and method to conduct a multi-party audio communication session
without the need for powerful servers or peers.
Inventors: |
Li; Jin (Sammamish, WA) |
Assignee: |
Microsoft Corporation (Redmond,
WA)
|
Family
ID: |
36046851 |
Appl.
No.: |
11/066,137 |
Filed: |
February 23, 2005 |
Prior Publication Data
|
|
|
|
Document
Identifier |
Publication Date |
|
US 20060187860 A1 |
Aug 24, 2006 |
|
Current U.S.
Class: |
370/267;
370/352 |
Current CPC
Class: |
H04L
65/4046 (20130101); H04L 67/1076 (20130101); H04L
67/104 (20130101); H04L 29/06027 (20130101); H04L
67/108 (20130101); H04L 69/04 (20130101); H04L
12/1813 (20130101) |
Current International
Class: |
H04L
12/16 (20060101); H04Q 11/00 (20060101); G06F
15/16 (20060101); G10L 21/00 (20060101); H04L
12/66 (20060101) |
Field of
Search: |
;370/260 |
References Cited
[Referenced By]
U.S. Patent Documents
Foreign Patent Documents
Other References
European Search Report, Application No. EP6100434, completed Mar.
28, 2006, received Apr. 10, 2006. cited by other .
Co-pending U.S. Appl. No. 10/887,406 entitled "Efficient
One-to-Many Content Distribution in a Peer-to-Peer Computer
Network" by J. Li, P. Chou, and C. Zhang, filed on Jul. 7, 2004.
cited by other .
G.722.1, "Coding at 24 and 32 kbit/s for hands-free operation in
systems with low frame loss". cited by other .
Lennox, J., Schulzrinne, H., "A protocol for reliable decentralized
conferencing", In Proc of 13th international workshop on network
and operating systems support for digital audio and video,
(NOSSDAV'2003), pp. 72-81, 2003, Monterey, California. cited by
other .
Li, J., Chou, P. A., and Zhang, C., "Mutualcast: An Efficient
Mechanism for Content Distribution in a Peer-to-Peer (P2P)
Network", MSR-TR-2004-100, Sep. 2004. cited by other .
Radenkovic, M., Greenhalgh, C., and Benford, S., "Deployment issues
for multi-user audio support in CVEs", In Proc. ACM Symp. On
Virtual Reality Software and Technology, pp. 179-185, 2002, Hong
Kong, China. cited by other .
Singh, K., Nair, G., and Schulzrinne, H., "Centralized Conferencing
using SIP", In Proc of the 2nd IP-Telephony Workshop, Apr. 2001.
cited by other.
|
Primary Examiner: Nguyen; Chau
Assistant Examiner: Chriss; Andrew
Attorney, Agent or Firm: Lyon & Harr, L.L.P. Fischer;
Craig S.
Claims
What is claimed is:
1. A method for a multi-party, real-time communication session
using peer nodes in a peer-to-peer computer network, comprising:
dividing an input stream into a plurality of frames; assigning each
frame to the peer nodes such that a peer node receives a number of
frames proportional to the available resources of the peer node;
and rotating mixing and redelivery of the plurality of frames on a
frame-by-frame basis among at least some of the peer nodes to
provide the multi-party real-time communication session.
2. The method as set forth in claim 1, wherein rotating mixing and
redelivery of the plurality of frames further comprises mixing and
redelivering at least part of the input stream in a round robin
manner.
3. The method as set forth in claim 2, wherein the input stream is
at least one of: (a) an arriving audio stream containing audio
content; (b) an arriving microphone signal containing audio
content, and further comprising mixing and redelivering the audio
content.
4. The method as set forth in claim 3, wherein mixing further
comprises: performing one of the following processes to generate
encoded audio coefficient rackets: (a) entropy decoding and inverse
quantizing of the arriving audio stream to generate encoded audio
coefficient packets; (b) transforming of the arriving microphone
signal into audio coefficient blocks and encoding the audio
coefficient blocks to generate encoded audio coefficient packets;
and combining the audio coefficient packets and blocks to generate
a resultant coefficient block that is a combination of the
coefficient packets and blocks for a certain frame.
5. The method as set forth in claim 4, further comprising:
quantizing the resultant coefficient block; and re-encoding the
quantized resultant coefficient block to generate a mixed audio
packet.
6. The method as set forth in claim 5, further comprising: decoding
the mixed audio packet; and inverse quantizing the decoded mixed
audio packet to obtain mixed transform coefficients; and applying
an inverse transform to the mixed transform coefficients to
generate an audio stream containing mixed audio content.
7. The method as set forth in claim 3, further comprising mixing
and redelivering audio content from each of the peer nodes except
from a certain peer node to which the audio stream and microphone
signal is sent.
8. The method as set forth in claim 3, further comprising: mixing
and redelivering at least one of the following: (a) the arriving
audio stream containing audio content; (b) the arriving microphone
signal containing audio content; and causing each peer node to
subtract its own audio content from a mixed audio content to reduce
echo.
9. The method as set forth in claim 1, further comprising: defining
a clique as the peer nodes that form a fully connected mesh; and
arranging the clique such that it serves as a super gateway node in
the multi-party, real-time communication session.
10. The method as set forth in claim 1, further comprising:
defining a clique as the peer nodes that form a fully connected
mesh; and arranging the clique such that it serves as a super
server in a star topology having additional nodes that act as
client nodes in the multi-party, real-time communication session
such that client nodes are exempted from the mixing task.
11. A computer-readable medium having computer-executable
instructions for performing the computer-implemented method recited
in claim 1.
12. A computer-implemented process for conducting a multi-party
real-time audio communication session between peer nodes in a
peer-to-peer computer network, each of the peer nodes having an
audio stream, comprising: a packet splitting step for splitting the
audio stream from each of the peer nodes into audio packets; an
audio mixing step for combining audio packets from a certain time
from each of the peer nodes to generate a mixed audio packet at
that time; and a rotation step for rotating audio mixing of mixed
audio packets and redelivery of the mixed audio packets on a
packet-by-packet basis among each of the peer nodes, such that at
least some of the peer nodes have a turn performing audio mixing
and redelivery.
13. The computer-implemented process as set forth in claim 12,
further comprising rotating the audio mixing and redelivery between
at least some of the peer nodes in a round robin manner.
14. The computer-implemented process as set forth in claim 13,
wherein the audio mixing step further comprises: encoding the audio
packets using an entropy encoder to generate entropy- encoded audio
packets; and mixing on a packet-by-packet basis the entropy-encoded
audio packets.
15. The computer-implemented process as set forth in claim 14,
further comprising: selecting entropy-encoded audio packets at the
certain time from each of the peer nodes; entropy decoding and
inverse quantizing the selected audio packets to generate
coefficient blocks corresponding to each of the selected audio
packets; combining the coefficient blocks to generate a resultant
coefficient block for each of the peer nodes at the certain time;
and quantizing and entropy re-encoding the resultant coefficient
block to produce the mixed audio packet.
16. The computer-implemented process as set forth in claim 12,
further comprising: entropy decoding and inverse quantizing the
mixed audio packets at a receiving peer node to obtain mixed
transform coefficients; and inverse modified discrete cosine
transforming the mixed transform coefficients to recover audio
content.
17. The computer-implemented process as set forth in claim 16,
wherein the audio content further comprises audio content from each
of the peer nodes except a source peer node that is performing
mixing and redelivery.
18. The computer-implemented process as set forth in claim 16,
wherein the audio content includes audio content from each of the
peer nodes, and further comprising having each of the peer nodes
subtract out its own audio content.
19. The computer-implemented process as set forth in claim 12,
further comprising: fully connecting the peer nodes to define a
clique; and organizing a network topology such that the clique
behaves as one of: (a) a super gateway node that interacts with
other nodes outside of the clique; (b) a super server such that the
clique is in a position of a server in a star network topology
interacting with client nodes outside of the clique.
20. A peer-to-peer audio communication system for engaging in a
multi-party real-time audio communication session between peer
nodes in a peer-to-peer network, comprising: an audio packet module
that divides an input audio stream having audio content into a
plurality of audio frames; an audio mixer that performs mixing of
the audio frames from each of the peer nodes at a particular frame
to produce a mixed audio packet at each of the plurality of audio
frames; a round robin processing technique that rotates audio
mixing and redelivery of the mixed audio content one at a time on a
frame-by-frame basis to each of the peer nodes in a round robin
manner; and an audio decoder that decodes the redelivered mixed
audio content to obtain audio for the audio communication session.
Description
TECHNICAL FIELD
The present invention relates in general to computer networking and
more particularly to a serverless peer-to-peer (P2P) multi-party
real-time audio communication system and method in which each of
the peers takes a turn mixing the audio and redelivering the
compressed audio, wherein the number of audio frames mixed and
redelivered by a certain peer is proportional to its available
resources (such as the upload bandwidth or computational
power).
BACKGROUND OF THE INVENTION
A multi-party audio communication system enables a group of people
to engage in a real-time audio communication session. In addition,
the system allows multiple people to be speaking at the same time.
Besides the audio components of the two-party audio communication
system (such as audio capture, acoustic echo cancellation (AEC),
automatic gain control (AGC), and audio/speech compression), the
multi-party audio communication system poses unique challenges in
audio mixing and network delivery.
By way of example, assume that n number of peer computers (or
peers) are engaged in a multi-party audio communication session,
with possible multiple concurrent speakers. Further assume that
each stream of audio requires a bandwidth of bw. The multi-party
audio communication system may be formed with a variety of
topologies and mixing strategies. One popular topology is a star
topology, as shown in FIG. 1A. A powerful central server, S,
receives audio streams from all peers (t.sub.1, t.sub.2, t.sub.3,
t.sub.4, and t.sub.5), mixes the audio streams, and sends the mixed
and re-encoded audio back to all peers.
The advantage of the star topology is that each peer uses the same
hardware as that of a two-party communication system, and thus
needs no modification. Only the server needs to be redesigned to
support a multi-party communication session. Consequently, the star
topology is a popular choice for commercial multi-party
communication solutions. One such system is set forth in a paper by
K. Singh, G. Nair, and H. Schulzrinne entitled "Centralized
Conferencing using SIP" in Proceedings of the 2.sup.nd IP Telephony
Workshop, April 2001. The main shortcoming of the start topology is
that a heavy computation and bandwidth burden is placed on the
server, S. The server, S, needs to receive n streams of compressed
audio (with nbw download bandwidth), decode, mix and re-encode
them, and send the mixed audio back to n peers (nbw upload
bandwidth).
A second common topology is a fully connected unicast network, as
shown in FIG. 1B. In a fully connected network, every peer is
connected to every other peer in the network. An example of this
type of topology is discussed in a paper by J. Lennox and H.
Schulzrinne entitled "A protocol for reliable decentralized
conferencing" in Proceedings of the 13.sup.th international
workshop on network and operating systems support for digital audio
and video, (NOSS-DAV'2003), pp. 72-81, 2003, Monterey, Calif. In
this topology, the peers (t.sub.1, t.sub.2, t.sub.3, t.sub.4, and
t.sub.5) do not perform any audio mixing or redelivery. Instead,
each speaker simply sends the compressed audio to every other peer.
In such a topology, each peer needs (n-1)bw upload bandwidth to
send the audio to the rest of the peer, and a maximum of (n-1)bw
download bandwidth to receive the incoming audio. One disadvantage
of this topology is the large increase in network traffic, which
places a large burden on each peer and the entire network.
A third possible topology is a generic graph that uses end system
mixing. An example of this type of topology is shown in FIG. 1C and
in a paper by M. Radenkovic, C. Greenhalgh, and S. Benford entitled
"Deployment issues for multi-user audio support in CVEs" in
Proceedings ACM Symposium on virtual reality software and
technology, pp. 179-185, 2002, Hong Kong, China. As shown in FIG.
1C, in this example peers a, b, f and g are leaf nodes, and do not
perform any mixing operations. The peers c, d and e serve as a
gateway node, which mixes and redelivers the audio for the nearby
peers. In general, a gateway node with m neighbors requires mbw
upload and download bandwidth to receive and redeliver the audio.
Since m is usually much smaller than n, the design of this topology
scales well to a large conferencing session. Nevertheless, the
disadvantage of this topology is that the burden on the gateway
node can be heavy. Another disadvantage is that as the chain of
gateways becomes long, the latency in audio delivery increases. Yet
another disadvantage is that the audio may also lose
synchronization along the chain of delivery.
A network level solution to further reduce the traffic in an audio
communication session is through IP multicast. In IP multicast, a
single packet that is transmitted from a source is duplicated at
routers along a distribution tree rooted at the source. In this
manner, content is delivered to an arbitrary number of receivers.
For example, in the star topology shown in FIG. 1A, a peer may
still send the compressed audio to the server via unicast. However,
the server, S, can multicast the mixed and re-encoded audio back to
n peers. A sample implementation of such system can be found in a
paper entitled "ConferenceXP: wireless classrooms, collaboration
and distance learning". The upload bandwidth of the server, S, is
reduced to bw.
One disadvantage, however, of IP multicast is that the requirement
on the download bandwidth of the server remains unchanged at nbw.
In the fully connected network shown in FIG. 1B, each speaker may
also multicast the compressed audio to every other peer in the
network. Again, the disadvantage of the IP multicast for the fully
connected network is that while the upload bandwidth of the peer is
reduced to bw, the download bandwidth of the peer remain unchanged
at (n-1)bw. Another disadvantage of IP multicast is that its
deployment is slow in the real world because of issues such as
inter-domain routing protocols, ISP business models (charging
models), congestion control along the distribution tree, and
security, among other things. As a result, except certain limited
university/corporate subnet and network test bed (such as
Internet2), native IP multicast support is not widely available.
Because of these problems in deploying a network-level multicast
service, the vast majority of traffic in the Internet today is
unicast based, whereby two computers directly talk to each
other.
One type of system and method for one-to-many content distribution
for file transfer over a P2P network is described in co-pending
patent application U.S. Ser. No. 10/887,406 entitled "Efficient
One-to-Many Content Distribution in a Peer-to-Peer Computer
Network" by J. Li, P. Chou, and C. Zhang, filed on Jul. 7, 2004.
However, that work involved one-to-many file transfer and
distribution, whereas an audio communication session involves
many-to-many distribution. In addition, that work made extensive
use of a TCP/IP queue. However, using a queue is impractical for
audio conferencing, because the packets must arrive in a timely
manner. Moreover, audio from different sources may be mixed, which
makes audio delivery unique in the audio communication
applications.
One disadvantage of existing multi-party audio communication
systems is that the mixing and redelivery role played by the peer
or server is fixed by the network topology. Another disadvantage of
existing audio communication systems is that they perform mixing
entire audio streams. Therefore, what is needed is an audio
communication system and method that makes the most efficient use
of network resources. Moreover, what is needed is a system and
method that avoid the disadvantages of the above-described network
topologies and is flexible in the mixing and redelivery roles
played by the peers. Further, what is needed is an audio
communication system and method that performs mixing on frames of
audio streams rather than the entire audio streams. Moreover, what
is needed is an audio communication system and method that avoid
the use of a queue and overcomes the delay problems of file
transfer techniques.
SUMMARY OF THE INVENTION
The invention disclosed herein includes a peer-to-peer (P2P) audio
communication system and method that provides a real-time
multi-party audio communication session for maximum efficiency. A
P2P network is a type of network in which each computer generally
has equivalent capabilities and responsibilities. The P2P audio
communication system and method divides compressed audio into
packets and sends each of the packets to a single peer for mixing
and redelivery. The number of audio packets mixed and redelivered
by a certain peer is proportional to its available resources. These
resources may include the upload bandwidth. Alternatively, it may
include the computation power. The P2P audio communication system
and method reduces the bandwidth required in a multi-party audio
communication sessions. Moreover, the P2P audio communication
system and method balances the audio serving load with the peer
upload bandwidths, and redistributes the cost of a multi-party
communication session among all participant peers in the P2P
network. This enables the P2P audio communication system and method
to conduct a multi-party audio communication session without the
need for powerful servers or peers.
Unlike prior techniques, where the mixing and redelivery is based
on the network topology, the P2P audio communication system and
method disclosed herein splits or divides the compressed audio into
packets or frames and lets each peer take a turn mixing and
redelivering the audio packets. The P2P audio communication system
and method flexibly balances the network bandwidth load of the
peers, such that peers having more resources are able to assist
those peers having fewer resources. In addition, through audio
mixing, the P2P audio communication system and method reduces the
bandwidth required to conduct a multi-party, real-time audio
communication session.
BRIEF DESCRIPTION OF THE DRAWINGS
The present invention can be further understood by reference to the
following description and attached drawings that illustrate aspects
of the invention. Other features and advantages will be apparent
from the following detailed description of the invention, taken in
conjunction with the accompanying drawings, which illustrate, by
way of example, the principles of the present invention.
Referring now to the drawings in which like reference numbers
represent corresponding parts throughout:
FIG. 1A is a block diagram illustrating a computer network having a
star topology.
FIG. 1B is a block diagram illustrating a computer network having a
fully connected unicast topology.
FIG. 1C is a block diagram illustrating a computer network having a
generic graph topology using end system mixing.
FIG. 2 is a block diagram illustrating an exemplary implementation
of the P2P audio communication system and method disclosed
herein.
FIG. 3 is a detailed block diagram illustrating a generic exemplary
implementation of the P2P audio communication system that is
contained on each of the peer nodes as shown in FIG. 2
FIG. 4 is a general flow diagram illustrating the general operation
of the P2P audio communication system shown in FIG. 3.
FIG. 5 is a detailed block/flow diagram of the operation of the
audio encoder used in this working example.
FIG. 6 is a detailed block/flow diagram of the operation of the
audio mixer used in this working example and shown in FIG. 3.
FIG. 7 is a detailed block/flow diagram of the operation of the
audio decoder used in this working example.
FIG. 8 illustrates the operation of the audio mixer shown in FIG. 6
for three peer nodes (1, 2 and 3) used in this working example.
FIG. 9 illustrates an exemplary example of a MutualCast clique of
the P2P audio communication system and method serving as a gateway
node in a generic multi-party conference graph.
FIG. 10 illustrates an exemplary example of a MutualCast clique of
the P2P audio communication system and method as a super server in
a multi-party communication session with a star topology.
FIG. 11 illustrates an example of a suitable computing system
environment in which the P2P audio communication system and method
shown in FIGS. 3 and 4 may be implemented.
DETAILED DESCRIPTION OF THE INVENTION
In the following description of the invention, reference is made to
the accompanying drawings, which form a part thereof, and in which
is shown by way of illustration a specific example whereby the
invention may be practiced. It is to be understood that other
embodiments may be utilized and structural changes may be made
without departing from the scope of the present invention.
I. Introduction
Although current one-to-many distribution techniques in
peer-to-peer (P2P) computer networks are more efficient than
directly sending content from source node to the peer nodes, these
techniques fail to achieve the most efficient content distribution
in the network. This is due to a number of factors. One factor is
that none of these current techniques adequately accounts for and
adapts to differences in bandwidth between the peer nodes. Another
factor is that these techniques fail to fully utilize the bandwidth
capabilities of all of the peer nodes on the network when
distributing the content.
The P2P audio communication system and method described herein is a
novel solution to conducting a multi-party audio communication
session. A key characteristic of the P2P audio communication system
and method is that the tasks of audio mixing and redelivering the
audio are rotated among the peers in a MutualCast clique. The
MutualCast clique includes a small number of peer nodes that form a
fully connected mesh. Using the special property of the waveform
coded audio that the audio mixing can be performed on the transform
domain and on a frame-by-frame basis, the P2P audio communication
system and method rotates the mixing and redelivering tasks among
the participant peers. This allows sharing of the network bandwidth
and computation load. Thus, the P2P audio communication system and
method can conduct a multi-party audio communication session
without a powerful server.
II. General Overview
FIG. 2 is a block diagram illustrating an exemplary implementation
of the P2P audio communication system and method disclosed herein.
It should be noted that FIG. 2 is merely one of several ways in
which the P2P audio communication system and method may implemented
and used.
Referring to FIG. 2, in this exemplary implementation a
fully-connected peer-to-peer (P2P) network 200 is shown. The P2P
network 200 that runs uses the P2P audio communication system and
method is also called a MutualCast clique. In this exemplary
implementation shown in FIG. 2, the MutualCast clique 200 includes
three peer nodes, namely peer node (1), peer node (2), and peer
node (3). The peer nodes (1), (2), (3), are fully connected, as
shown by the arrows. Each of the peer nodes (1), (2), (3), contains
the P2P audio communication system and method.
FIG. 3 is a detailed block diagram illustrating a generic exemplary
implementation of the P2P audio communication system 300 that is
contained on each of the peer nodes as shown in FIG. 2. In general,
referring to FIG. 3, the P2P audio communication system 300
receives network audio from each of the peers and microphone input
from a local peer, mixes the audio, and output the mixed audio to
the peer nodes as well as the local peer. The local peer node mixes
and redelivers the mixed audio. At least some of the peer nodes
take turns performing the mixing and redelivery tasks in a round
robin manner.
Specifically, FIG. 3 illustrates the P2P audio communication system
300 on the local peer node (not shown). Assume that there are N
peer nodes in the P2P network. Input to the system 300 includes
network audio from peer (1) 310 to peer (N) 315. The ellipsis in
FIG. 3 indicate that not all peer nodes are shown. In addition,
microphone streams 320 also are input to the system 300. The
microphone streams 320 are from one or more microphones at the
local peer. The P2P audio communication system 300 includes an
audio mixer 330 that divides the input audio into frames or packets
and mixes at a certain frame each of the packets.
The encoded network audio from peer (1) 310 is processed by
performing entropy decoding and inverse quantizing (1) 340. As
explained in detail below, this partially decodes the encoded
network audio from peer (1) 310 and generates blocks of MDCT
transform coefficients. Similarly, the encoded network audio from
each of the peer to peer (N) 315 are processed by performing
entropy decoding and inverse quantizing (N) 345. Once again, the
ellipsis in FIG. 3 indicates that not all entropy decoding and
inverse quantizing is shown. The microphone streams are processed
using a modified discrete cosine transform (MDCT) module 350 to
also produce MDCT transform coefficients.
The audio mixer is used to mix the audio content from the network
audio from peers (1) to (N) and the microphone streams 320. This
produces mixed audio packets. This mixed audio content is
redelivered as mixed audio to peer (1) 360, as mixed audio to peer
(N), and as mixed audio for the other peers, as shown by the
ellipsis.
The delivered network audio from peer(1) to peer(N) are also mixed
for local playback. The mixed audio, minus the local microphone
input content (if local input has been mixed by other peers), then
are fed into an inverse MDCT module 370. This recovers the audio
content of the network peers and allows playback through the
speaker 380.
It should be noted that the audio from other peers is encoded prior
to being transmitted over the P2P network. In addition, once the
mixed audio content is received by a peer node, that peer decodes
the mixed audio to recover the audio associated with the audio
communication session. Thus, each of the peer nodes contain an
audio mixer 330. In addition, the audio mixer 330 further contains
multiple audio encoder and audio decoder components. How and when
each of these components is used depends on the processing being
done by the peer at any given time. For example, when the local
peer is performing mixing and redelivery, the audio mixer 330
receives one frame of coded audio from each peer in the Mutualcast
clique, performs a partial decoding, mixes the audio, performs a
partial encoding and sends one frame of mixed audio to each network
peer. When the local peer becomes a client peer and the mixing and
redelivery are assigned to another peer, the audio mixer 330 simply
sends one frame of an encoded microphone stream input to the peer
in charge of mixing, receives one frame from the mixing peer,
performs a decoding operation, and plays back the mixed audio
locally. Switching between components as needed occurs quickly,
because the round robin scheme used by the P2P audio communication
system and method changes the functionality of the local peer node
quickly in a round robin manner. The details of the audio mixer
330, audio encoder, and audio decoder are described below.
III. Operational Overview
The operation of the P2P audio communication system 300 shown in
FIG. 3 now will be discussed. FIG. 4 is a general flow diagram
illustrating the general operation of the P2P audio communication
system 300 shown in FIG. 3. In general, the P2P audio communication
method divides an input audio stream into frames or packets and
rotates the processing of the frames among the peer nodes in the
P2P network. More specifically, the method begins by dividing an
input audio stream into a plurality of frames (box 400). Next, one
of the peer nodes in the P2P network is selected (box 410).
The selected peer node is assigned to process frames proportional
to the available resources of the selected node (box 420). In other
words, a peer node having a large amount of resources is assigned a
greater number of frames, while a peer node having a fewer amount
of resources is assigned a fewer number of frames. Each peer node
takes a turn processing the frames. Processing includes the mixing
of audio content in the input audio stream and redelivery of the
audio to the peer nodes. Thus, the selected peer node performs
mixing of audio frames and redelivers the mixed audio frames to
other peer nodes in the P2P network. The processing of the frames
is rotated among the peer nodes (box 430). In a preferred
embodiment, this rotation is performed in a round robin manner or
fashion, such that each peer node takes a turn.
IV. Operational Details and Working Example
The details of the operation of the P2P audio communication system
and method shown in FIGS. 3 and 4 now will be discussed. In order
to more fully understand the P2P audio communication system and
method disclosed herein, the operational details of an exemplary
working example are presented. It should be noted that this working
example is only one way in which the P2P audio communication system
and method may be implemented.
Audio Content
Mixing of Waveform Coded Audio
In this working example, the audio of the P2P audio communication
system and method was encoded with a waveform codec. Such a
waveform codec is Siren/G.722.1, described in G.722.1, "Coding at
24 and 32 kbit/s for hands-free operation in systems with low frame
loss." An alternative codec is MP3, described in Scot Hacker press,
"MP3: The definitive Guide". A further alternative codec is Windows
Media Audio, in Microsoft Press, "Inside Windows Media". FIG. 5 is
a detailed block/flow diagram of the operation of the audio encoder
500 used in this working example.
Referring to FIG. 5, the audio input 510 containing audio waveforms
first was split or divided into frames. In this working example,
each frame was approximately 20 millisecond (ms) long. It should be
noted that because there was a 50 % overlap in between frames, the
total algorithmic delay is 40 ms, which doubles that of the frame.
Nevertheless, each quantized and entropy coded frame coefficients
was still 20 ms long. Next, each frame was transformed by a
modified discrete cosine transform (MDCT) module 520 into
coefficient blocks (C.sub.i,j). The subscript i indicates the peer
and the subscript j indicates the frame number. Thus, in FIG. 5,
the coefficient block C.sub.i,1 indicates the first frame for the
i.sup.th peer and the coefficient block C.sub.i,2 indicates the
second frame for the i.sup.th peer. Next, the coefficient blocks
were sent to a quantizer 530 and quantized. The quantized
coefficient blocks then were entropy encoded by the entropy encoder
540 into packets p.sub.i,j. In FIG. 5, the encoded packets
P.sub.i,1 and P.sub.i,2 indicate encoded packets for the first and
second frames of the i.sup.th peer.
Since the MDCT is a linear operation, the waveform-coded audio can
not only be mixed in the transform domain, but also be mixed on a
frame-by-frame basis. FIG. 6 is a detailed block/flow diagram of
the operation of the audio mixer 600 used in this working example.
In FIG. 6, the mixing of a certain frame of two compressed audio
packets is shown. In general, the audio mixer 600 decodes the
compressed audio packets of a certain frame to obtain coefficients
blocks, combines the coefficient blocks, and re-encodes the blocks
to obtain a single compressed audio packet for a frame of audio.
Rather performing the mixing operation on an entire input audio
stream, as in done in current audio conferencing systems, the audio
mixing technique of the P2P audio communication system and method
mixes frames of the audio.
In particular, in this working example the compressed audio packet
P.sub.1,j was decoded using an entropy decoder (1) 610 and an
inverse quantizer (1) 620 to obtain a coefficient block for a
certain peer. Similarly, the compressed audio packet P.sub.2,j was
decoded using an entropy decoder (2) 630 and an inverse quantizer
(2) 640 to obtain a coefficient block for a another peer. The
resultant coefficients were MDCT transform coefficients. These MDCT
transform coefficients then were added together or combined using
the combination module 650, as shown by the "+" sign in the
combination module 650. It should be noted that although a "+" sign
is shown in FIG. 6, for audio, combination may be subtraction (a
minus sign "-") or addition (a plus sign "+") and there will be no
audible difference for the coded audio. Next, the resultant
coefficient block was quantized using a mixing quantizer 660 and
then entropy re-encoded using a mixing entropy encoder 670. This
generated a frame containing the mixed audio packet,
P.sub.1,j+P.sub.2,j. No audio packets of any other frames were
accessed during the mixing process.
FIG. 7 is a detailed block/flow diagram of the operation of the
audio decoder 700 used in this working example. At a receiver, the
mixed packets were decoded normally. Referring to FIG. 7, the mixed
audio packet P.sub.1,j+P.sub.2,j was input to the audio decoder
700. Each frame was processed by the entropy decoder (3) 710 and
then by the inverse quantizer (3) 720. This produced mixed MDCT
transform coefficients, C.sub.1,1+C.sub.2,1 and
C.sub.1,2+C.sub.2,2. Next, the mixed MDCT transform coefficients
were processed using the inverse MDCT module 730. The resultant
output was the mixed audio waveform from peer 1 and 2 (audio
1+audio 2) 740.
Round-Robin Rotational Mixing and Redelivery of Audio Content
Another component in this working example was the rotational mixing
and redelivery of the audio content in a round robin manner. In
other words, each of the peers take a turn in mixing and
redelivering the audio content. In this working example, a
MutualCast clique was used that consisted of a small number of peer
nodes that formed a fully connected mesh. Using the property that
the waveform-coded audio content can be mixed on a frame-by-frame
basis, the P2P audio conferencing system and method rotated the
mixing and redelivery operation among the peers. This ensured that
the bandwidth and computation load were appropriately distributed
in the most efficient manner. In a preferred embodiment, the
rotation was performed in a round robin manner or fashion such that
each peer node took a turn mixing and redelivering the audio.
FIG. 8 illustrates an audio mixing session for three peer nodes (1,
2 and 3) used in this working example. An audio mixing schedule is
shown at the bottom of FIG. 8. Referring to the audio mixing
schedule, the peer nodes 1, 2 and 3 were in charge of audio mixing
and redelivery at frames 3k, 3k+1 and 3k+2, respectively. Thus, at
the 1.sup.st frame, peer 2 mixed and redelivered the audio packets.
While peer 2 was in charge of mixing and redelivery, peers 1 and 3
sent their coded audio P.sub.1,1 and P.sub.3,1 to peer 2, as shown
in FIG. 8. The incoming audio packets then were entropy decoded and
inverse quantized back to the MDCT coefficients C.sub.1,1 and
C.sub.3,1. Peer 2 then added its own coefficients C.sub.2,1. The
mixed audio then was sent back to peers 1 and 3.
In order to avoid echo, the source audio was not mixed and sent
back. In other words, referring to FIG. 8, peer 2 added together
coefficients C.sub.1,1 and C.sub.2,1, quantized and entropy encoded
the sum of the coefficients. Peer 2 then sent the mixed packet
P.sub.1,1+P.sub.2,1 back to peer 3. Similarly, the mixed packet
P.sub.3,1+P.sub.2,1 was sent by peer 2 to peer 1. At the
destination, the mixed audio packets from the different peers were
sorted, entropy decoded, inverse quantized and inverse MDCT
transformed for play back.
Referring again to the mixing schedule of FIG. 8, at the 2.sup.nd
frame, peer 3 assumed the mixing role. Peers 1 and 2 sent their
compressed audio packets at the 2.sup.nd frame P.sub.1,2 and
P.sub.2,2 to peer 3. Peer 3 then mixed the incoming audio packets
with its own coefficients C.sub.3,2. Next, peer 3 sent the mixed
packet P.sub.3,2+P.sub.2,2 to peer 1 and the mixed packet
P.sub.3,2+P.sub.1,2 to peer 2.
At the 3.sup.rd frame, peer node 1 became the mixing node, and so
forth, as shown in FIG. 8. By time sharing the mixing and
redelivery tasks, the bandwidth and computational cost of the
mixing is distributed among each of the peers. As a result, a group
of less powerful peers can conduct a multi-party audio
communication session without need for a server.
In the P2P audio communication system and method, a MutualCast
clique consists of n nodes. Each peer node sends and receives
2(n-1) packets every n frames. Among them, (n-1) packets are sent
and received during the n-1 frames that it does not perform the
mixing operation. Moreover, (n-1) packets are sent and received
during the frame that it performs the mixing and redelivery
operation. The upload/download bandwidth required is thus
(2-2/n)bw. It may also be calculated that on average, (2-2/n)
streams of audio are decoded and re-encoded by each peer. During
the mixing, the peer performs (n-1) entropy decoding and inverse
quantization operations, and (n-1) forward quantization and entropy
encoding operations.
There are at least two possibilities as to the audio content
contained in the redelivered mixed audio packets. In a preferred
embodiment, the mixed audio packet does not contain the source (or
selected) peer nodes' audio content. In other words, the rotational
audio mixing technique includes not mixing and not sending back a
certain peer's source audio. In an alternate embodiment, the mixed
audio packet contains audio content from each of the peer nodes and
it is up to each peer node to subtract out its own audio content
from the mixed audio packet. For example, the mixed audio includes
packets (m.sub.j=p.sub.1j+p.sub.2,j+p.sub.3,j) for frame j. Then,
the same mixed audio is sent back to all peers. In order to reduce
or eliminate echo, each peer is responsible for subtracting Its own
audio from the mixed audio. For example, peer i would subtract
p.sub.ij from m.sub.j, which is a mixing operation with subtraction
instead of addition. The advantage of this alternate embodiment is
that the peer only needs to perform a single forward quantization
and entropy encoding operation during the mixing. Moreover, if IP
multicast is supported among all the peers, the mixing peer may
also multicast the mixed packet to the rest of the peers. The
disadvantage of the alternate embodiment is that since the mixed
audio needs to be quantized and entropy coded, the component audio
p.sub.i,j in the mixed packet m.sub.j is different from the audio
p.sub.i,j hold by the peer i. Thus, residual echo may persist. This
residual echo is more obvious with the increase of the number of
the peers and/or the decrease in the coding bitrate of the mixed
audio. Thus, due to this residual echo problem, the preferred
embodiment is to not mix all of the audio packets.
Allocation of Mixing Tasks
The P2P audio communication system and method allocates the mixing
and redelivery tasks of the audio on a frame-by-frame basis. In
this manner, the P2P audio communication system and method assigns
more mixing tasks to peers having greater resources, and fewer
mixing tasks to peers having fewer resources. The paramount
resource considered by the P2P audio communication system and
method is the upload bandwidths of the peers. In increasingly
common networks, the total upload bandwidths of the P2P network are
much smaller than the total download bandwidths. This is especially
true for end-user nodes on the cable modem and ADSL networks, for
which the balance is asymmetrically skewed towards larger download
bandwidth. Even for the user nodes on the university/corporate
networks, the download bandwidth can still be larger than the
available upload bandwidth as the user caps the upload bandwidth.
Therefore, it is advantageous to allocate more mixing and
redelivery tasks to the peer with higher available upload
bandwidth, and fewer tasks to the peer with lower upload
bandwidth.
A second resource considered by the P2P audio communication system
and method is the peak upload bandwidth (or the physical link
bandwidth) of the peer. During the mixing, a peer of the P2P audio
communication system and method receives and sends out (n-1) audio
packets to (n-1) peers. The traffic characteristics of a peer in
the P2P audio communication system is bursty. It is helpful to
assign more mixing and redelivery tasks to the peer with a faster
physical link, or the peer that is connected to the router with a
relatively large token bucket, so that the delay caused by sending
packets to multiple peers can be reduced.
Normally, the download bandwidths and the computation resources of
the peers are not a bottleneck. Nevertheless, the P2P audio
communication system and method may also take these into
consideration in the allocation as well. It should be noted that if
the slower or less powerful nodes are allowed to deliver fewer
packets, they become leeches of the faster, more powerful peers.
Whether to allow such leech behavior is a design choice between
better audio communication performance verses fairness on
contribution.
Delay
The maximum delay of the P2P audio communication system and method
now will be calculated and presented. Let the network transmission
delay between peer nodes i and j be d.sub.i,j. Assuming that the
delay caused by the mixing operation is negligible, the delay of
peer i to receive an audio frame mixed by peer k amounts to:
.noteq..times..times. ##EQU00001##
The maximum delay of peer i can be calculated as:
.noteq..times..noteq..times..times. ##EQU00002## while the maximum
delay of the P2P audio communication system and method is:
.times..times..noteq..times..times. ##EQU00003## which is two times
the network delay of the farthest peer pair in a MutualCast clique.
V. Super Gateways and Super Servers
A MutualCast clique of the P2P audio communication system must be
formed by a set of fully connected peer nodes. As shown above, the
delay increases as the clique grows large. Accordingly, the number
of nodes in a MutualCast clique ideally should not be very large. A
reasonable number is between 3 to 7. Nevertheless, the MutualCast
clique can serve as a super gateway or super server, and thus
function in a much larger network, while retaining all the
functionality of the P2P audio communication system and method
described herein.
Super Gateway Node
The P2P audio communication system can function as a super gateway
node. This allows the MutualCast clique to serve as a "super"
gateway node in a large multi-party communication session. In this
situation, the rest of the nodes form a generic graph and use end
system mixing. FIG. 9 illustrates an exemplary example of a
MutualCast clique of the P2P audio communication system and method
serving as a gateway node in a generic multi-party communication
session. This configuration is particularly well-suited for a
multi-party communication session having a large number of peer
nodes, where a small number of close-by nodes are fully connected
and form a MutualCast clique.
As shown in the exemplary example of FIG. 9, the MutualCast clique
900 formed by the nodes a, b and c serves as a "super" gateway node
for nodes d, e, f, g and i. Each peer node in the MutualCast clique
900 serves as a gateway for the nodes attached. For example, peer
node a is attached to two nodes d and e outside of the MutualCast
clique 900. Thus, within the MutualCast clique 900, node a merges
the audio of d and e with its own, and delivers the combined audio
(a+d+e) to nodes b and c using the P2P audio communication system
and method. At frames where the node a is mixing for the MutualCast
clique 900, the combined audio (a+d+e) is mixed with the input from
the node b (audio b+i) and sent to node c. Likewise, the combined
audio (a+d+e) is mixed with the input from node c (audio c+f+g+h)
and sent to node b. Node a also mixes the inputs from nodes b and
c, combined it with its own, m=a+b+i+c+f+g+h. (4) and sends m+d to
node e, and sends m+e to node d.
At frames where node a is not mixing, the combined audio (a+d+e) is
sent to the mixing node (b or c) at the moment. Node a also
receives the mixed input from nodes b or c, combines them with its
own to form the mixed frame m, and sends m+d to node e, and sends
m+e to node d.
Super Server
The P2P audio communication system also can function as a super
server. In this situation, the remainder of the nodes are the
client nodes in a star topology, and the MutualCast clique may
consist as fewer as two nodes. FIG. 10 illustrates an exemplary
example of a MutualCast clique as a super server in a multi-party
communication session with a star topology.
As shown in FIG. 10, peer nodes a and b form a two-node MutualCast
clique 1000, which serves as a super server for the rest client
nodes c, d, e, f and g. The peer nodes in the MutualCast clique
1000 again form a fully connected mesh. In addition, each client
node outside is connected to all peer nodes in the MutualCast
clique 1000. This configuration is more suited for small-to-medium
size networks (such as between 4 and 16), where there are a few
powerful broadband nodes that share to serve as the server.
Another scenario involved this configuration occurs when there is a
network address translator (NAT) or firewall. Specifically, the
client nodes may be behind NAT/firewall. They can connect to the
nodes that are directly connected to the Internet, in other words,
those of the MutualCast clique 1000. However, they can not connect
to each other. The mixing and redelivery operation of such network
is very similar to the P2P audio communication system and method
described above. The only difference is that in this embodiment the
client nodes are exempted from the mixing and redelivery tasks. At
each frame, one peer of the MutualCast clique 1000 mixes and
redelivers the audio packets for the rest of the peers, both inside
and outside the MutualCast clique 1000.
VI. Bandwidth and Computation Load Analysis
In this section, the bandwidth requirement of different audio
communication session scenarios using the P2P audio communication
system and method is calculated and compared to those scenarios not
using the P2P audio communication system and method. First,
consider an n-party communication session. If all peers are of
equal bandwidth, the P2P audio communication system and method
requires the upload/download bandwidth of (2-2/n)bw for each and
every peer node. In particular, for a three-node MutualCast clique,
the bandwidth required is 1.34 bw. The same three-node
communication session needs a node with bandwidth at least 2 bw to
conduct a multi-party communication session with either star
topology, or generic graph. Thus, the P2P audio communication
system and method can conduct multi-party communication sessions
even when all peer nodes are less resourceful.
Second, consider the case when the MutualCast clique serves as a
super gateway in a large graph. Let the gateway node be connected
to m nodes. Normally, the gateway node needs mbw upload and
download bandwidth to mix and redeliver the audio traffic. By
replacing the gateway node with an n-node MutualCast clique, and
assuming m/n nodes are attached to each node, the upload/download
bandwidth requirement of each node is reduced in this super gateway
to:
.times. ##EQU00004##
By way of example, let m=6 and n=3. In this case, the P2P audio
communication system and method reduces the bandwidth requirement
from 6 bw to 3.34 bw. The use of the MutualCast clique as a super
gateway also may reduce the bandwidth requirement of the gateway
nodes.
Finally, consider the case when the MutualCast clique is used as a
super server. Assume again that there are m clients. Without using
the P2P audio communication system and method, the server needs mbw
upload and download bandwidth to serve the m clients. However, when
using the P2P audio communication system and method and a
MutualCast clique of n nodes, it can be calculated again that on
average, each peer node in the MutualCast clique needs only
.times. ##EQU00005## upload/download bandwidth. As an example, let
m=5 and n=2. The P2P audio communication system and method reduces
the bandwidth requirement of the server node from 5 bw to 3.5 bw.
It can be easily deduced that MutualCast reduces the computation
load of the peer, the super gateway node and the super server node
by the same proportion as well. VII. Exemplary Operating
Environment
The P2P audio communication system and method are designed to
operate in a computing environment and on a computing device. The
computing environment in which the P2P audio communication system
and method operates will now be discussed. The following discussion
is intended to provide a brief, general description of a suitable
computing environment in which the P2P audio communication system
and method may be implemented.
FIG. 12 illustrates an example of a suitable computing system
environment in which the P2P audio communication system and method
shown in FIGS. 3 and 4 may be implemented. The computing system
environment 1100 is only one example of a suitable computing
environment and is not intended to suggest any limitation as to the
scope of use or functionality of the invention. Neither should the
computing environment 1100 be interpreted as having any dependency
or requirement relating to any one or combination of components
illustrated in the exemplary operating environment 1100.
The P2P audio communication system and method is operational with
numerous other general purpose or special purpose computing system
environments or configurations. Examples of well known computing
systems, environments, and/or configurations that may be suitable
for use with the P2P audio communication system and method include,
but are not limited to, personal computers, server computers,
hand-held, laptop or mobile computer or communications devices such
as cell phones and PDA's, multiprocessor systems,
microprocessor-based systems, set top boxes, programmable consumer
electronics, network PCs, minicomputers, mainframe computers,
distributed computing environments that include any of the above
systems or devices, and the like.
The P2P audio communication system and method may be described in
the general context of computer-executable instructions, such as
program modules, being executed by a computer. Generally, program
modules include routines, programs, objects, components, data
structures, etc., that perform particular tasks or implement
particular abstract data types. The P2P audio communication system
and method may also be practiced in distributed computing
environments where tasks are performed by remote processing devices
that are linked through a communications network. In a distributed
computing environment, program modules may be located in both local
and remote computer storage media including memory storage devices.
With reference to FIG. 11, an exemplary system for implementing the
P2P audio communication system and method includes a
general-purpose computing device in the form of a computer 1110.
The peer nodes shown in FIG. 11 are examples of the computer
1110.
Components of the computer 1110 may include, but are not limited
to, a processing unit 1120, a system memory 1130, and a system bus
1121 that couples various system components including the system
memory to the processing unit 1120. The system bus 1121 may be any
of several types of bus structures including a memory bus or memory
controller, a peripheral bus, and a local bus using any of a
variety of bus architectures. By way of example, and not
limitation, such architectures include Industry Standard
Architecture (ISA) bus, Micro Channel Architecture (MCA) bus,
Enhanced ISA (EISA) bus, Video Electronics Standards Association
(VESA) local bus, and Peripheral Component Interconnect (PCI) bus
also known as Mezzanine bus.
The computer 1110 typically includes a variety of computer readable
media. Computer readable media can be any available media that can
be accessed by the computer 1110 and includes both volatile and
nonvolatile media, removable and non-removable media. By way of
example, and not limitation, computer readable media may comprise
computer storage media. Computer storage media includes volatile
and nonvolatile removable and non-removable media implemented in
any method or technology for storage of information such as
computer readable instructions, data structures, program modules or
other data.
Computer storage media includes, but is not limited to, RAM, ROM,
EEPROM, flash memory or other memory technology, CD-ROM, digital
versatile disks (DVD) or other optical disk storage, magnetic
cassettes, magnetic tape, magnetic disk storage or other magnetic
storage devices, or any other medium which can be used to store the
desired information and which can be accessed by the computer
1110.
The system memory 1130 includes computer storage media in the form
of volatile and/or nonvolatile memory such as read only memory
(ROM) 1131 and random access memory (RAM) 1132. A basic
input/output system 1133 (BIOS), containing the basic routines that
help to transfer information between elements within the computer
1110, such as during start-up, is typically stored in ROM 1131. RAM
1132 typically contains data and/or program modules that are
immediately accessible to and/or presently being operated on by
processing unit 1120. By way of example, and not limitation, FIG.
11 illustrates operating system 1134, application programs 1135,
other program modules 1136, and program data 1137.
The computer 1110 may also include other removable/non-removable,
volatile/nonvolatile computer storage media. By way of example
only, FIG. 11 illustrates a hard disk drive 1141 that reads from or
writes to non-removable, nonvolatile magnetic media, a magnetic
disk drive 1151 that reads from or writes to a removable,
nonvolatile magnetic disk 1152, and an optical disk drive 1155 that
reads from or writes to a removable, nonvolatile optical disk 1156
such as a CD ROM or other optical media.
Other removable/non-removable, volatile/nonvolatile computer
storage media that can be used in the exemplary operating
environment include, but are not limited to, magnetic tape
cassettes, flash memory cards, digital versatile disks, digital
video tape, solid state RAM, solid state ROM, and the like. The
hard disk drive 1141 is typically connected to the system bus 1121
through a non-removable memory interface such as interface 1140,
and magnetic disk drive 1151 and optical disk drive 1155 are
typically connected to the system bus 1121 by a removable memory
interface, such as interface 1150.
The drives and their associated computer storage media discussed
above and illustrated in FIG. 11, provide storage of computer
readable instructions, data structures, program modules and other
data for the computer 1110. In FIG. 11, for example, hard disk
drive 1141 is illustrated as storing operating system 1144,
application programs 1145, other program modules 1146, and program
data 1147. Note that these components can either be the same as or
different from operating system 1134, application programs 1135,
other program modules 1136, and program data 1137. Operating system
1144, application programs 1145, other program modules 1146, and
program data 1147 are given different numbers here to illustrate
that, at a minimum, they are different copies. A user may enter
commands and information into the computer 1110 through input
devices such as a keyboard 1162 and pointing device 1161, commonly
referred to as a mouse, trackball or touch pad.
Other input devices (not shown) may include a microphone, joystick,
game pad, satellite dish, scanner, radio receiver, or a television
or broadcast video receiver, or the like. These and other input
devices are often connected to the processing unit 1120 through a
user input interface 1160 that is coupled to the system bus 1121,
but may be connected by other interface and bus structures, such
as, for example, a parallel port, game port or a universal serial
bus (USB). A monitor 1191 or other type of display device is also
connected to the system bus 1121 via an interface, such as a video
interface 1190. In addition to the monitor, computers may also
include other peripheral output devices such as speakers 1197 and
printer 1196, which may be connected through an output peripheral
interface 1195.
The computer 1110 may operate in a networked environment using
logical connections to one or more remote computers, such as a
remote computer 1180. The remote computer 1180 may be a personal
computer, a server, a router, a network PC, a peer device or other
common network node, and typically includes many or all of the
elements described above relative to the computer 1110, although
only a memory storage device 1181 has been illustrated in FIG. 11.
The logical connections depicted in FIG. 11 include a local area
network (LAN) 1171 and a wide area network (WAN) 1173, but may also
include other networks. Such networking environments are
commonplace in offices, enterprise-wide computer networks,
intranets and the Internet.
When used in a LAN networking environment, the computer 1110 is
connected to the LAN 1171 through a network interface or adapter
1170. When used in a WAN networking environment, the computer 1110
typically includes a modem 1172 or other means for establishing
communications over the WAN 1173, such as the Internet. The modem
1172, which may be internal or external, may be connected to the
system bus 1121 via the user input interface 1160, or other
appropriate mechanism. In a networked environment, program modules
depicted relative to the computer 1110, or portions thereof, may be
stored in the remote memory storage device. By way of example, and
not limitation, FIG. 11 illustrates remote application programs
1185 as residing on memory device 1181. It will be appreciated that
the network connections shown are exemplary and other means of
establishing a communications link between the computers may be
used.
The foregoing description of the invention has been presented for
the purposes of illustration and description. It is not intended to
be exhaustive or to limit the invention to the precise form
disclosed. Many modifications and variations are possible in light
of the above teaching. It is intended that the scope of the
invention be limited not by this detailed description of the
invention, but rather by the claims appended hereto.
* * * * *